Voice over internet protocol telephone system and method

ABSTRACT

An Internet Protocol telephone system and method uses a telephone (26) to place and receive voice over Internet Protocol (VoIP)-based telephone calls and public switched telephone network (PSTN)-based telephone calls. An off-hook condition with the telephone (26) is detected and a sequence of signals generated by the telephone (26) is received. At least a first signal generated by the telephone (26) is buffered while the system attempts to detect a predetermined signal that signifies a VoIP-based call. Upon detection of the predetermined signal, the system intercepts subsequent signals in the sequence, absent the at least first signal that was buffered, and places the VoIP-based call via an internet (12). Otherwise, the system places the PSTN-based call via a PSTN (16).

FIELD OF THE INVENTION

The present invention relates to a voice over Internet Protocoltelephone system and method.

BACKGROUND OF THE INVENTION

Most current systems for Internet Protocol (IP) telephony or voice overan Internet Protocol (VoIP) are difficult and sometimes impractical touse. Since these systems are internet-based, they typically require theuser to utilize his or her personal computer (PC) to connect to aninternet server in order to place and receive internet-based calls.These PCs sometimes have a telephone connected to them, but often theuser is left using the PC's speakers and microphone for the telephoneconversation. Using the PC's speakers and microphone for such use isawkward and limits current user acceptability.

A solution to avoid using the PC to place and receive internet-basedcalls is to provide the user with a custom made telephone that supportsVoIP-based telephone calls. A problem with this solution is that itrequires the user to purchase additional telephone equipment to supportthe VoIP capabilities. As such, the user is forced into purchasingredundant telephone hardware equipment.

Thus, a need exists for a system and method that enables users to placeand receive internet-based calls via the user's existing telephoneequipment operating in its current fashion.

BRIEF DESCRIPTION OF THE DRAWINGS

A preferred embodiment of the present invention is now described, by wayof example only, with reference to the accompanying drawings in which:

FIG. 1 illustrates a whole-home Internet Protocol (IP) telephone systemusing a network premises gateway according to the preferred embodimentof the present invention;

FIG. 2 illustrates the whole-home IP telephone system using a secondaryinternet access device and service provider to access the internet in analternative embodiment of the present invention;

FIG. 3 illustrates the network premises gateway of FIG. 1 connected to apublic switched telephone network and an in-premises plain old telephoneservice network according to the preferred embodiment of the presentinvention;

FIG. 4 illustrates a hardware block diagram of the network premisesgateway of FIG. 1 according to the preferred embodiment of the presentinvention;

FIG. 5 illustrates a hardware block diagram of a telephony subsystem,which is a component of the hardware block diagram of FIG. 4, accordingto the preferred embodiment of the present invention; and

FIG. 6 illustrates a hardware block diagram of an IP telephony engine,which is a component of the hardware block diagram of FIG. 4, accordingto the preferred embodiment of the present invention.

DETAILED DESCRIPTION OF A PREFERRED EMBODIMENT

The present invention implements a voice over Internet Protocol (IP)telephone system and method, suitable for whole-home and other uses,which enables customers to place and receive internet-based calls (i.e.,voice over Internet Protocol (VoIP)) via the customer's existing plainold telephone service (POTS) telephone equipment operating in itscurrent fashion. Such a configuration allows a user to utilize hisexisting POTS telephones to place and receive "standard" public switchedtelephone network (PSTN)-based calls as well as VoIP-based calls, thuspreventing the user from purchasing redundant telephone hardwareequipment.

The preferred embodiment of the present invention supplies customerswith broadband internet access in their home or business via a networkpremises gateway 10 as shown in FIG. 1. The whole-home IP telephonesystem with VoIP functionality and associated internet connectivity isembedded in the network premises gateway 10, thus allowing the networkpremises gateway 10 to enable access to the wide area network (WAN) andthe internet 12.

Alternatively, customers may also access the WAN and the internet 12 viaanother provider or service inside their home or business as shown inFIG. 2. In an alternative embodiment of the present invention, thenetwork premises gateway 10 relies on the WAN and the internet 12connectivity from an external and independent internet access device 14(i.e., an integrated services digital network (ISDN) modem, a digitalsubscribe line (xDSL) modem, cable modem, etc.).

In the preferred embodiment, the network premises gateway 10 connects toa PSTN 16 via a PSTN network interface unit (NIU) 18. The PSTN NIU 18 istypically found on the outside of most homes in the United States. Thisis the demarcation point between the customer's equipment and thetelephone company's equipment.

As shown in FIG. 3, the network premises gateway 10 also connects to anin-premises POTS network 20 via a RJ-41 type interface 22. The RJ-41type interface 22 allows the network premises gateway 10 to arbitratethe in-premises POTS network 20 between "standard" PSTN-based calls andVoIP-based calls to and from the WAN and the internet 12.

The primary interfaces or networks in the network premises gateway 10are the in-premises POTS network 20, the PSTN 16 and the broadbandconnection to the WAN and the internet 12. FIG. 3 illustrates anenlarged view of the primary interfaces and their associated connectionsused to connect the network premises gateway 10 to the PSTN 16 and thein-premises POTS network 20.

As shown in FIG. 3, the connection from the in-premises POTS network 20leads to a "break out" box, a "fan out" box or a splitter 24 for thein-premises POTS network 20. Such a connection allows all the POTStelephones 26 in the premises to be accessible via the network premisesgateway 10. It should be noted that the network premises gateway 10implemented in the present invention can be configured to supportmultiple telephone lines for customers who currently have more than oneanalog telephone line in their home or business.

Further, it is important to note that the addition of a POTS cordlesstelephone 28 operates with and supports the features of the presentinvention. An analog-based POTS cordless telephone 28 has a wirelessinterface for connecting the cordless handset to an analog base or basestation. The analog base or base station connects the POTS cordlesstelephone 28 to the in-premises POTS network 20 and to the networkpremises gateway 10. An existing POTS cordless telephone 28 may bedigital over the air (e.g., a 900 MHz telephone), but the associatedbase station has a standard analog interface. Thus, the presentinvention does not preclude a POTS cordless telephone 28 from operatingin its normal fashion.

FIG. 4 illustrates a high level hardware block diagram of the networkpremises gateway 10. In the block diagram, there is a connection to theWAN and the internet 12 and support for a wireless network of wirelessdigital handsets 30 or devices (additional advantages of supporting thewireless network of wireless handsets 30 or devices are described indetail below). The components of interest in FIG. 4 are the systemcontroller and memory component 32 and the telephony subsystem 34 andthe IP telephony H.323 engine 36, which are described in detail below.

The system controller and memory component 32 controls the functions andoperation of the network premises gateway 10. User interaction with thenetwork premises gateway 10 is controlled by a user interface programwhich resides in and is executed on the system controller and memorycomponent 32. The system controller and memory component 32 is designedto contain functions and operate in a manner similar to a standardmicroprocessor controlled computer system. The system controller andmemory component 32 is the heart of the network premises gateway 10 andit controls all the network premises gateway's 10 functions, operations,and states.

The system controller and memory component 32 comprises a memory system(not shown), a comparator (not shown) and a microprocessor (not shown).The memory system buffers at least a first signal generated by the POTStelephone 26. The comparator attempts to detect a predetermined signalthat signifies a VoIP-based call. The microprocessor and associatedsoftware program intercepts subsequent signals in the sequence, absentthe at least first signal that was buffered, and places the VoIP-basedcall via an internet when the predetermined signal is detected.

FIG. 5 graphically illustrates the telephony subsystem 34. The telephonysubsystem 34 comprises a telephony manager 38, a POTS interface 40 and atelephony crossbar 42. The telephony subsystem 34 also comprises othercomponents, such as an IP telephony interface 44, a POTS output port 46(used for connection of a local POTS telephone 26 to the networkpremises gateway 10), a compression-decompression engine (CODEC) andpacketizer 48 and a music on-hold module 50. Although the telephonysubsystem 34 comprises a multitude of components, the followingdescription is limited to the telephony manager 38, the POTS interface40 and the telephony crossbar 42.

The telephony manager 38 supplies all the functions for VoIP-based callswhich are normally supplied by a switch in the PSTN 16 for POTStelephones 26. For example, the telephony manager 38 comprises a dualtone multi-frequency (DTMF) detection and call progress generator 52.The DTMF detection and call progress generator 52 comprises a detectorfor receiving a sequence of signals generated by the POTS telephone. Inaddition to performing DTMF detection, the DTMF detection and callprogress generator 52 generates DTMF signaling and supplies dial tonesand other appropriate call progress tones for the POTS telephones 26when the network premises gateway 10 is operating. The incoming callhandler 54 generates ringing for the POTS telephones 26 when signaled bythe system controller and memory component 32.

Other features of the telephony manager 38 include: support for controlof the telephony crossbar 42 via a crossbar manager 56 (described indetail below); DTMF generation and pulse dialing, flash hook,on/off-hook; and basic user interface for control of user interactionfor VoIP answering and origination.

The POTS interface 40 contains the RJ-41 type interface 22 forconnecting the network premises gateway 10 to the in-premises POTSnetwork 20 and the PSTN NIU 18. The POTS interface 40 comprises a statedetector for detecting an off-hook condition with the POTS telephone 26.The POTS interface 40 contains analog-to-digital converters (ADCs--whichare not shown) for encoding the analog signals from the POTS telephones26 to a digital sixty-four (64) kilobits per second pulse codemodulation (PCM) for the telephony crossbar 42 and from analog signalsvia the PSTN 16 to the telephony crossbar 42. Likewise, the POTSinterface 40 contains digital-to-analog converters (DACs--which are notshown) for decoding the digital sixty-four (64) kilobits per second(Kbps) PCM signals received from the telephony crossbar 42 for analogdelivery to the POTS telephones 26 and from the telephony crossbar 42for analog delivery to the PSTN 16. It should be noted that there is aset of DAC and ADC for each connection, i.e., to the in-premises POTSnetwork 20 and to the PSTN 16 via the PSTN NIU 18.

The telephony crossbar 42 is the "spine" of the telephony subsystem 34;the telephony crossbar 42 couples the telephony manager 38 and the POTSinterface 40 to each other. The telephony crossbar 42 is also a routerfor all telephony calls, PSTN and VoIP, alike. The telephony crossbar 42routes the digitally encoded, sixty-four (64) Kbps PCM, audio signals toand from the POTS interface 40, IP telephony H.323 engine 36 andtelephony manager 38 components at the direction of the crossbar manager56.

The crossbar manager 56 contains an application specific integratedcircuit (ASIC) for controlling the state and operation of the telephonycrossbar 42. It should be noted that the ASIC could be replaced with abus controller which is made up of an embedded microprocessor, memoryand associated software residing on a single chip. The crossbar manager56 maintains the data flow between the submodules of the telephonysubsystem 34 across the telephony crossbar 42.

The crossbar manager 56 further has the ability to mute the signals andto inject music from a digital audio stream produced at the musicon-hold synthesizer 50 for on-hold calls. Since the crossbar manager 56contains input for the music on-hold module 50 for creating music, theuser has the ability to place remote parties from the telephony call,either PSTN-based or VoIP-based, on hold, mute the signals and injectmusic from a digital audio stream produced at the music on-hold module50. At the control of the user via the user interface, the systemcontroller and memory component 32 enables the music on-hold module 50and instructs the crossbar manager 56 to replace the sixty-four (64)Kbps PCM audio samples from the in-premises POTS network 20 withsixty-four (64) Kbps PCM audio samples from the music on-hold module 50.Such a replacement causes internal audio from the POTS telephones 26 tobe muted (since these samples are no longer sent to the PSTN 16 or theinternet 12 (for VoIP-based calls)) and replaced with music. Thus, thetelephony crossbar 42 links the on-hold call with the music on-holdmodule 50 of the telephony subsystem 34, and sends this music to theremote party instead of audio digitized from the in-premises POTStelephones 26 or other telephony devices in the premises. In the eventthat the user prefers just muting the on-hold telephone call withoutmusic, the system controller and memory component 32 instructs thecrossbar manager 56 to "drop" or disregard the sixty-four (64) Kbps PCMsamples from the in-premises POTS network 20.

The telephony crossbar 42 can also combine audio signals from varioussources for call conferencing, including conferencing PSTN-based callswith VoIP-based calls. The telephony crossbar 42 combines calls bysending multiple digital sixty-four (64) Kbps PCM streams to a commonDAC in the POTS interface 40. Each DAC in the POTS interface 40 hasassociated circuitry (not shown) which sums the digital sixty-four (64)Kbps PCM streams into a single combined digital stream and automaticallyadjusts the single combined digital stream, via an automatic gaincontrol (AGC) (also not shown), before the single combined digitalstream enters the DAC. The AGC guarantees that the single combineddigital stream remains within the dynamic range of the DAC. Sendingmultiple digital sixty-four (64) Kbps PCM streams to a common DAC causesaudio from two telephone calls, VoIP based and PSTN-based, to be "added"together or combined, thus allowing call conferencing between multiplesources.

The IP telephony H.323 engine 36 is a standard H.323 engine forsupporting VoIP-based calls. The IP telephony H.323 engine 36 integratesthe in-premises POTS telephones 26 to the broadband WAN and internet 12connection which allows the POTS telephones 26 to place and receiveVoIP-based calls. Likewise, the IP telephony H.323 engine 36 may be usedto support wired or wireless IP devices 30 which support VoIPfunctionality (i.e., wireless IP devices which have a microphone,speaker and dialing pad) within the premises for PSTN-based andVoIP-based calls.

The IP telephony H.323 engine 36 converts sixty-four (64) Kbps PCMsampled, POTS audio signals into H.323 compliant audio streams for VoIPfunctionality. As graphically shown in FIG. 6, a G.723.1 audio CODEC toPCM 58, 60, telephony manager interface 62, networking interface 64,H.225 multiplex & demultiplex subsections 66, 68 and H.245 signaling andcontrol subsections 70, 72 are the components for the IP telephony H.323engine 36. Two blocks are shown in the figure for the components andfunctions described above, i.e., H.225 multiplex & demultiplexsubsections 66, 68 and H.245 signaling and control subsections 70, 72.Two of each of these components are shown and described to demonstratethe support of multiple VoIP calls in one network premises gateway 10.There is a software program executing on the system controller andmemory 32 which performs the H.323 standard-based functions forVoIP-based call setup and teardown, including but not limited tostandard Q.931 signaling for call setup and initiate. The softwareprogram executing on the system controller and memory 32 controls thetelephony manager interface 62 for managing, at the H.245 subsections70, 72, and audio and stream control to provide service overnon-guaranteed links (e.g., transmission control protocol (TCP)/IP).G.723.1 audio CODEC to PCM 62 performs the transcoding from sixty-four(64) Kbps PCM to 5.3 or 6.3 Kbps low bit rate audio.

In operation, the DTMF detection and call progress generator 52 performsDTMF functions on signals generated from a POTS telephone 26, sends adigital representation of the information to the system controller and-memory component 32 for buffering into memory, and replaces the PSTNdial tone on all POTS telephones 26 with a slightly modified dial tone(i.e., the audible characteristics, tone and/or pitch of the PSTN dialtone is altered) when a POTS telephone 26 is taken off-hook. Theslightly modified dial tone reminds the user that he has the option ofplacing an internet-based call, thus indicating that the networkpremises gateway 10 is currently on-line. Should the network premisesgateway 10 be shut-off or down, the user hears the PSTN supplied dialtone when a POTS telephone 26 is taken off-hook.

The telephony manager 38 also comprises an incoming call handler 54. Theincoming call handler 54 supports PSTN call waiting notification duringthe presence VoIP calls and ring detection and generation with cadenceinformation to the user. For example, the incoming call handler 54signals the system controller and memory component 32 and DTMF detectionand call process generator 52 which notifies the user of an incomingPSTN-based call when the present call is a VoIP-based call. The systemcontroller and memory 32 is alerted via a standard H.323 alert messageof an incoming VoIP call and signals the DTMF detection and call processgenerator 52 which notifies the user of 1) an incoming VoIP-based callwhen the present call is also a VoIP-based call; and 2) an incomingVoIP-based call when the present call is a PSTN-based call. Notifyingthe user of an incoming PSTN-based call when the present call is also aPSTN-based call is currently supported by the local telephone company.

In operation, during the presence of a call, the DTMF detection and callprocess generator 52 notifies the user of the incoming call by anaudible tone that is user configurable, so that the user can ascertainwhether the call is a PSTN-based call or a VoIP-based call. When no callis present, i.e., the POTS phones are on-hook, an incoming PSTN-basedcall and an incoming VoIP-based call is preferably programmed to havedifferent ringing cadence, thus informing the user whether the incomingcall is a PSTN-based or VoIP-based call based solely on the ringingcadence.

The system control and memory component 32 in forms the user of anincoming PSTN-based or VoIP-based call by transmitting the following:caller identification information (discussed in detail below); anelectronic mail (email) message to a known user configurationaddress(es); or a telephony page from the PSTN 16 to a standard pager.

The system controller and memory component 32 is configurable, based oninformation from the incoming call handler 54 or receipt of H.323alerts, to notify the user of an incoming electronic mail message, anincoming VoIP-based facsimile and an incoming PSTN-based facsimile inthe same manner described above with respect to incoming PSTN-based andVoIP-based calls. For example, email message alerts are sent by havingthe remove email mailbox ping the network premises gateway 10, thesystem controller and memory component 32 would activate a program inthe system and notify the incoming caller handler 54 on receipt of theping from the email messaging system. The program or set of operationsat the system controller and memory component 32 would instruct the callhandler 54 to ring, with a special cadence, the POTS telephone 26notifying the user of an email message arriving at the user's remoteemail mailbox. Internet mailboxes typically conform to known standards,e.g., simple mail transfer protocol (SMTP), post office protocol (POP),internet message access protocol (IMAP), etc. Most of these standardssupport notification based on receipt of email messages. In the eventthat a user has a non-standard mailbox, other systems can be designedthat periodically check the mailbox and then send notification to theincoming call handler 54 based on the detection of new email messages inthe mailbox. This assumes that a program resides on the systemcontroller and memory component 32 for periodically checking whether newmail has arrived. This program is similar to the systems which arereadily available to customers today, such as, biff and rbiff (UNIXdaemons), PCBIFF (biff for Windows PCs), macbiff (biff for Macintoshsystems) and the Windows PC program check mail.

Alternatively, the user configures the system so that the systemcontroller and memory component 32 instructs the DTMF detection and callprocess generator 52 to use the PSTN 16 to send a "standard"telephony-based page on receipt of an email message. Once the systemcontroller and memory component 32 detects the presence of new mail inthe users mailbox a local program can instruct the DTMF detection andcall progress generator 52 to initiate a new call to the PSTN 16. Thiscall is to a user configured telephone number of a pager and then thesystem controller and memory component 32 sends pertinent, userconfigured, information to the pager. The information sent to the pagermay include, but is not limited to, the sender of the email message, thesubject/title of the message, and/or the message body.

With respect to email messages, the system controller and memorycomponent 32 provides the following additional options to notify theuser of an incoming email message: translating the content of the emailmessage into a voice message for retrieval inside the premises (i.e., atany of the POTS telephones 26) or via dial up from a remote location(i.e., from any PSTN connection); or converting the content of the emailmessage into a facsimile. Similar to the interface discussed with pagingor ringing, the system controller and memory component 32 contains aprogram which integrates a text-to-speech processor (not shown butcontained as software in the system controller and memory component 32in a manner which computers generate speech today) to enable reading ofthe email message over the POTS telephones 26. For example, notificationof the email message pings the system controller and memory component 32and the system controller and memory component 32 instructs the incomingcall handler 54 to ring the POTS telephone 26 (with a special cadence).If the user picks up the POTS telephone 26, the system controller andmemory component 32 reads the email message to the user over the POTStelephone 26. It should be noted that a speech-to-text processor (notshown) could be integrated to perform the opposite function,speech-to-email message, from any POTS telephone 26 in the premises.

For example, the whole-home IP telephone system has memory, answers theincoming telephone calls, and stores messages from the calling party inthe system memory, either PSTN-based or VoIP-based, in the event that noone is able to pick up the POTS telephone 26 or answer call with adigital handset 30 in the premises. In other words, the whole-home IPtelephone system is integrated in a manner so that it operates as ananswering machine for both PSTN-based and VoIP-based calls alike. Once aremote caller leaves a message on the system, the system controller andmemory component 32 may be configured to access the WAN and internet 12and send email messages based on the voice message that the callingparty left. The email message is stored in the system memory of thesystem controller and memory component 32. Using speak-to-textcapabilities which are readily available today, the message is encodedinto text and included with the aforementioned email message.

Likewise, the network premises gateway 10 can be configured by the userto send a page notification based on a message that is left on ananswering machine by the calling party. It is important to note that thevoice mailbox is the same for both VoIP-based messages and "standard"voice messages from the PSTN. Sending a page notification based on amessage that is left on an answering machine by the calling partyoperates just as paging on receipt of an email message.

With respect to facsimiles, either PSTN-based or VoIP-based, an opticalcharacter recognizer (either as a software system contained in thesystem controller and memory component 32 or as dedicated hardware,i.e., an ASIC) is integrated into the incoming call handler 54 toprovide the following further options to notify the user of an incomingfacsimile: translating via optical character recognition into a textmessage and further into a voice message for retrieval inside thepremises or via a dial up from a remote location; or translating opticalcharacter recognition into a text message and further into an emailmessage.

The incoming call handler 54 also supports caller identification in thesame format that is known in the art. Caller identification informationis passed as a 1200 baud, seven (7) data bits, one (1) stop bit encodeddata stream for sending calling party information across the PSTNnetwork for receipt and display at caller identification compatibledevices. An example of a formatted output for caller identificationcompatible device as known in the art is the following:

Date--Feb. 28

Time--1:34 PM

Number--(407) 555-1111

Since the network premises gateway 10 implemented in the presentinvention interfaces between the in-premises POTS network 20 and the WANand internet 12, the network premises gateway 10 receives VoIP-basedcalls and translates IP address information into a format that iscompatible with the caller identification format above. Most of theVoIP, H.323-based calls contain more detail in their call stateinformation (as will be described later), however, there are cases andtimes when the user wants to have a calling party's machine name lookedup from the standard domain name services (DNS) available on theinternet 12. In such a situation, look-ups for domain naming system issupported by containing a software system on the system controller andmemory component 32 similar to nslookup, a standard UNIX command. Forexample, nslookup on "207.25.71.29" yields:

% nslookup 207.25.71.29

Name Server: argus.cso.uiuc.edu

Address: 128.174.5.58

Name: www9.cnn.com

Address: 207.25.71.29

In this example, the system controller and memory component 32 accessesthe internet 12 and supplies the name "www9.cnn.com". The user is ableto ascertain that the calling party is located at one of the CNN hypertext transfer protocol (http) servers. Since the called party in theVoIP scenario may only receive the IP address of the calling party,nslookup generates the name and the "standard" caller identificationformat is used to transmit the data to the in-premises calleridentification device. The system controller and memory component 32instructs the system to convert the data gathered from the DNS lookup toanalog instructions compatible with standard caller identificationfunctions. The system controller and memory component 32 instructs theDTMF detection and call progress generator 52 to translate theinformation to the analog modem codes, used for caller identification,and then the system controller and memory component 32 instructs theincoming call handler 54 to send the information on the POTS network 20to compatible caller identification devices.

Further, more information is supplied by the IP telephony H.323 engine36, which is received from the incoming H.323 VoIP-based call. Thisinformation is provided during the H.225.0 and Q.931 signaling andmessaging functions as called for in the H.323 standard. For example,the IP telephony H.323 engine 36 has the ability to send additionalsubscriber (calling party) information (e.g., name, location, personalinformation, such as "finger" information). This information istranslated and sent on the POTS network 20 and to compatible calleridentification devices in the manner discussed above.

For the case where the calling party only allows the IP telephony H.323engine 36 to distribute/broadcast the address of the originating call,the network premises gateway 10 performs a nslookup, or equivalent, andreferences the name and IP address against a database contained in thesystem controller and memory component 32 of the network premisesgateway 10 in order to do a name look-up.

After the translation, the caller identification information is sentover the in-premises POTS network 20, at the time of ringing the POTStelephones 26 for an incoming VoIP call, in the standard and knownanalog caller identification format which is supported by calleridentification systems available on the market today. This makesexisting caller identification boxes and POTS telephones 26, which haveintegrated caller identification functions, compatible with the VoIPsystem.

In operation, when placing a call, the network premises gateway 10 viathe DTMF detection and call progress generator 52 detects an off-hookcondition with a POTS telephone 26. The DTMF detection and call progressgenerator 52 receives a sequence of signals generated by the POTStelephone 26 and buffers at least the first signal generated by thetelephone. This technique is identical to that described in Newlin etal., U.S. patent application Ser. No. 08/735,295 filed Oct. 22, 1996,entitled Apparatus, Method and System for Multimedia Control andCommunication, Motorola Docket Number PD05688AM. The DTMF detection andcall progress generator 52 attempts to detect a sequence ofpredetermined signals call (e.g., the signals generated from the "#"keys of the POTS telephone 26) that signify a "non-standard" PSTN-basedcall, in this case a VoIP-based call. It should be noted that,alternatively, a single predetermined signal can signify a"non-standard" PSTN-based call as well.

When the sequence of predetermined signals is detected, the networkpremises gateway 10 enters a VoIP mode, intercepts subsequent signals inthe sequence, absent the signals buffered, and places the VoIP-basedcall via the internet. Once in the VoIP mode, the network premisesgateway 10 supports some form of dialing capability so that the user candial an IP number directly. For example, the user could dial"192#98#18#83" which would signify the IP number "192.93.18.83". Thesystem controller and memory component 32 interacts with the DTMFdetection and call progress generator 52 to detect the numbers dialedand control H.323 engine for placing the internet-based call. Thenetwork premises gateway 10 may also support voice (computer-generated)menuing in order to make selecting of called users easier. All functionsof the user interface would be controlled by the system controller andmemory component 32.

When the sequence of predetermined signals is not detected, the networkpremises gateway 10 enters a POTS mode and transmits the sequence ofsignals, including the signals buffered, to the PSTN 16. The userinteracts with the POTS telephone 26 and the PSTN 16 as normal.

As mentioned briefly previously, the addition of a wireless digitaldevice or handset 30 (e.g., a personal digital assistant or any otherdigital device including a simple dedicated digital telephony handset)which supports telephony features via its own user interface andconnects via a wireless link to the network premises gateway 10 providesthe same features as above along with additional features to the presentinvention. Fundamentally, the present invention provides systemintegration to use a wireless digital handset for PSTN-based andVoIP-based calls via a common interface without having a separate CODECexclusively for the VoIP-based calls.

The handset 30 is digital and the digital data is converted, at thenetwork premises gateway 10, from sixty-four (64) Kbps PCM to low bitrate data for VoIP functionality without going to the analog domain. Forexample, when placing an outgoing call from the digital wireless handset30, the digital wireless handset 30 receives a type-of-call selectionand a sequence of signals representative of a telephone number to placethe outgoing call following the type-of-call selection from a user.Analog signals (e.g., an electrical signal from a microphone generatedby the speaker's voice) are converted to digital signals at the digitalwireless handset 30 and transmitted to a network premises gateway 10.The digital signals are translated to a format compatible for a networkused in completing the outgoing call at the network premises gateway 10,wherein the network is the PSTN 16 for PSTN-based calls and an internet12 for VoIP-based calls. When receiving an incoming call at the digitalwireless handset 30, digital signals are transmitted from the networkpremises gateway 10 to the digital wireless handset 30 and converted toanalog signals at the digital wireless handset 30 in order to bebroadcasted to the user via the digital wireless handset's speaker(s).Cost and complexity reduction is inherent to this design. The H.323coding, i.e., call setup and management, is still performed at thenetwork premises gateway 10 and does not need to be reproduced in thehandset; however, the CODEC at the wireless digital handset 30 is usedto convert between the analog and digital domains.

Integrating the wireless digital handset 30 which operates as atelephony cordless telephone with the WAN and internet 12 connectionallows a user to access the configuration of the network premisesgateway 10, and thus the characteristics of their WAN and internetconnection 12, from their wireless digital handset 30. Moreover, asstated above, the user can access email messages from the wirelessdigital handset 30 without, however, going off-hook on the in-premisesPOTS network 20.

Caller identification could be provided at the wireless digital handset30 for VoIP calls as described above as well. With a wireless digitalhandset 30, however, caller identification information would not have tobe converted into analog telephone signals, in other words, aproprietary digital caller identification could be utilized for thedigital handsets. The caller identification information is sent directlyto the wireless digital handset 30 via the IP protocol.

The whole-home IP telephone system is configured in a manner so that ifthe network premises gateway 10 is shut off or in a crashed state, thein-premises POTS network 20 operates exactly as if the network premisesgateway 10 was not installed. Such a configuration allows the RJ-41interface 22 and the interface to pass PSTN-based calls on the PSTNconnection 16 to ensure that PSTN-based calls are placed and receivedeven when the network premises gateway 10 is disconnected. Thus, thepresent invention maintains the reliability and availability of the PSTN16.

While the invention has been described in conjunction with a specificembodiment thereof, additional advantages and modifications will readilyoccur to those skilled in the art. Moreover, even though the presentinvention has been designed to support the H.323 standard for VoIPfunctionality, the network premises gateway 10 can also be designed tosupport other, possibly proprietary, VoIP functionality techniques. Theinvention, in its broader aspects, is therefore not limited to thespecific details, representative apparatus and illustrative examplesshown and described. Various alterations, modifications and variationswill be apparent to those skilled in the art in light of the foregoingdescription. Thus, it should be understood that the invention is notlimited by the foregoing description, but embraces all such alterations,modifications and variations in accordance with the spirit and scope ofthe appended claims.

We claim:
 1. In an Internet Protocol telephone system using a telephoneto place and receive voice over Internet Protocol (VoIP)-based andpublic switched telephone network (PSTN)-based telephone calls, a methodcomprising:detecting a presence of an incoming telephone call; detectingan off-hook condition from the telephone; receiving a sequence ofsignals generated by the telephone; buffering at least a first signalgenerated by the telephone; attempting to detect a predetermined signalthat signifies a VoIP-based call; intercepting subsequent signals in thesequence, absent the at least first signal that was buffered, andfacilitating the VoIP-based call via an internet when the predeterminedsignal is detected; and signaling, via and audible tone, a user of thetelephone during the VoIP-based call of the presence of a telephonecall, the audible tone having a first ringing cadence when the telephonecall is a VoIP-based call and the audible tone having a second ringlingcadence when the telephone call is a PSTN-based call.
 2. The methodaccording to claim 1 wherein the incoming telephone call is a VoIP-basedcall.
 3. The method according to claim 1 wherein the incoming telephonecall is a PSTN-based call.
 4. The method according to claim 1 furthercomprising ringing the telephone via the internet when an electronicmail message arrives at a mailbox.
 5. The method according to claim 1further comprising, when the predetermined signal is not detected thatsignifies a VoIP-based call:determining that a telephone call is beingmade; transmitting the sequence of signals generated by the telephone toa PSTN; detecting a presence of an incoming VoIP-based call havingInternet Protocol address information; and signaling a user of thetelephone during the telephone call of the presence of the incomingVoIP-based call.
 6. The method according to claim 5 furthercomprising:translating the Internet Protocol address information into aformat compatible for caller identification to create VoIP calleridentification information; and presenting to the user of the telephonethe VoIP caller identification information.
 7. The method according toclaim 6 wherein the step of presenting to the user of the telephoneoccurs at the time of signaling the user of the telephone during thetelephone call of the presence of the incoming VoIP-based call.
 8. Themethod according to claim 1 wherein the telephone is cordless.